search for: maxexpirey

...ery 30 minutes with a cron job. I contacted Grandstream and although they didn't have a specific answer, they alluded to the fact that lack of registration might be the cause. I performed the following: Increased the "Register Expiration" in the device to 60 minutes Increased "maxexpirey" and "defaultexpirey" to 3800 seconds Configured the device with a static IP address Added defaultip=172.16.1.123 and host=dynamic to sip.conf It has now been running for 48 hours without a single lockup. Here are some of the settings that I am using: In Grandstream: Register Exp...

I have a handful of 7912's connected to my asterisk 1.2.14 server. (6 to be exact). I get the X on the display sometimes for loosing registration. I have the config file for the 7912's SipRegInterval: 60 and asterisk is the default. ; maxexpirey=3600 ;defaultexpirey=120 I've not changed them. How can I keep these phones online and stop loosing registration? Thanks, Jerry

...lagged users to re-register quickly. - check from time to time all users but no too often to see if is logged and can be called. Overall i want only lagged users to reregister and users with good response time to be check from time to time. defaultexpiry = 900 defaultexpirey = 900 maxexpiry = 300 maxexpirey = 300 minexpiry = 60 registerattempts = 5 registertimeout = 5 rtpholdtimeout = 900 rtptimeout = 60 jbmaxsize = 60 jbresyncthreshold = 200 qualify = yes qualify = 600 qualifyfreq = 60 Thank you. P.S. If you consider that i use too much options you can tell me what to drop. I use asterisk 1.8.6.0....

...NETWORK address localmask = 255.255.255.0 ; Internal netmask context = default ; Default for incoming calls ;srvlookup = yes ; Enable SRV lookups on outbound calls ;pedantic = yes ; Enable slow, pedantic checking for Pingtel ;tos=lowdelay ;tos=184 ;maxexpirey=3600 ; Max length of incoming registration we allow ;defaultexpirey=120 ; Default length of incoming/outoing registration ;notifymimetype=text/plain ; Allow overriding of mime type in NOTIFY ;videosupport=yes ; Turn on support for SIP video disallow=all...

...le to get the SIP channel working with this warning as well, but it took a lot more RELOADs. Any ideas? SIP.conf ====== [general] ; context=incoming-bogus-calls bindport=5060 ; Port to bind to (SIP is 5060) bindaddr=0.0.0.0 ; Address to bind to (all addresses on machine) maxexpirey=3600 ; Must be larger than the re-register timeout on the router defaultexpirey=3600 notifymimetype=text/plain rtptimeout=60 rtpholdtimeout=300 disallow=all allow=ulaw ; ; This section is because i'm behind nat ; register=>6477235412:<mypassword>@sip.unlimitel.ca/647...

Hi, Is it possible to use asterisk with Free World Dialup (FWD) ? Did someone manage to make it work? how? Best, -P -- __________________________________________________________ Sign-up for your own personalized E-mail at Mail.com http://www.mail.com/?sr=signup CareerBuilder.com has over 400,000 jobs. Be smarter about your job search http://corp.mail.com/careers

...can not register to Voipwise with Trixbox. It is always in "unregistered" state in sip registry. Here is my last sip trunk configuration: PEER DETAILS: allow=g729 bindport=5060 disallow=alldtmfmode=rfc2833 fromdomain=sip.voipwise.com fromuser=username host=sip.voipwise.com insecure=very maxexpirey=120 pickupgroup=1 port=5060 secret=pass type=peer username=username Register string: username:pass at sip.voipwise.com <username%3Apass at sip.voipwise.com> Do you have any suggestions? Thank you. Mert -------------- next part -------------- An HTML attachment was scrubbed... URL: http://l...

...NE@sip.broadvoice.com/PHONE [Broadvoice] type=friend username=PHONE authuser=PHONE fromuser=PHONE secret=secret host=sip.broadvoice.com port=5060 context=default fromdomain=sip.broadvoice.com canreinvite=no dtmfmode=inband insecure=very permit=sip.broadvoice.com qualify=yes disallow=all allow=ulaw maxexpirey=180 defaultexpirey=160 videosupport=no exten => _9XXXXXXX,1,Dial(SIP/Broadvoice/${EXTEN:1},${SMVOICE_DIAL_LONG_TIMEOUT},mtT) exten => _91XXXXXXXXXX,1,Dial(SIP/Broadvoice/${EXTEN:1},${SMVOICE_DIAL_LONG_TIMEOUT},mtT)

...er. Stanaphone has assured that their preferred codec is ulaw.... This is what I have in sip.conf: [general] port = 5060 ; Port to bind to bindaddr = 0.0.0.0 ; Address to bind to externip = frode.dyndns.org localnet = 192.168.0.0/16 context=stighess tos=lowdelay maxexpirey=3600 ; Max length of incoming registration we allow defaultexpirey=60 ; Default length of incoming/outoing registratio bandwidth=high disallow=all allow=ulaw ....... [stanaphone] type=friend username=91438xxxx fromuser=91438xxxxx secret=********* host=sip.stanaphone.com contex...

....conf' file: ;************************************************************* ; ; SIP Configuration for Asterisk ; [general] port = 5060 ; Port to bind to bindaddr = 0.0.0.0 ; Address to bind to context = default ; Default for incoming calls tos=184 maxexpirey=3600 ; Max length of incoming registration we allow defaultexpirey=120 ; Default length of incoming/outoing registration disallow=all ; Disallow all codecs allow=ulaw ; Allow codecs in order of preference allow=alaw [1004] type=friend...

Hello, I am trying to REGISTER asterisk to a SIP server, which is listening on Port 6060 (not 5060). The sip.conf file contains register=18474201111:quintum@192.168.2.94:6060/18474201111 maxexpirey=3600 defaultexpirey=120 But the REGISTER message is sent to Port 6060, but the Request-URI still contains, 5060. This is being rejected by SIP server. REGISTER sip:192.168.2.94 SIP/2.0 The request line instead should be like this: REGISTER sip:192.168.2.94:6060 SIP/2.0 See the...

...; Enable slow, pedantic checking for Pingtel ; and multiline formatted headers for strict ; SIP compatibility ;tos=lowdelay ; IP QoS parameter, either keyword or value ; like tos=184 ;maxexpirey=3600 ; Max length of incoming registration we allow realm=asterisk ; Our global authentication realm ;defaultexpirey=120 ; Default length of incoming/outoing registration ;notifymimetype=text/plain ; Allow overriding of mime type in NOTIFY ;videosupp...

I'm getting these messages every 7-10 seconds. -- Registered SIP '532' at x.x.x.x port 52956 expires 60 -- Registered SIP '532' at x.x.x.x port 56988 expires 60 -- Registered SIP '529' at x.x.x.x port 51444 expires 60 -- Registered SIP '529' at x.x.x.x port 64044 expires 60 -- Registered SIP '532' at x.x.x.x port 52956 expires 60 -- Registered SIP

...lso fixed the IP address of the BudgetTone. When I receive a call on my Asterisk, it would ring my FXS as before. However, after I pick up, it hangs within a few seconds (Hungup Zap1-1 in the log). The configuration I have in * is the following: sip.conf ----------- [general] port=5060 context=sip maxexpirey=3600 defaultexpirey=60 disallow=all allow=ulaw allow=gsm [1000] contet=sip type=friend username=1000 secret=????? (not the real one) host=dynamic mailbox=1000 canreinvite=yes dtmfmode=rfc2833 I did not change the above configuration when I moved the budgetTone from the LAN to the Internet (Wan)....

...ming calls port=5060 ; UDP Port to bind to (SIP standard port is 5060) bindaddr=0.0.0.0 ; IP address to bind to (0.0.0.0 binds to all) srvlookup=yes ; Enable DNS SRV lookups on outbound calls tos=lowdelay ; lowdelay,throughput,reliability,mincost,none maxexpirey=7200 ; Max length of incoming registration we allow defaultexpirey=3600 ; Default length of incoming/outoing registration videosupport=yes ; Turn on support for SIP video disallow=all ; First disallow all codecs allow=ulaw ; Allow codecs in order of prefer...

...onfirmed that this is dependent on the order in the conf file, not numeric order) sip.conf :- [general] port = 5060 bindaddr = 0.0.0.0 pedantic = no autocreatepeer = no context = sip registertimeout=20 localnet = 10.10.10.0/255.255.255.0 srvlookup = yes tos=0xb8 rtptimeout=300 rtpholdtimeout=1800 maxexpirey=3600 defaultexpirey=1200 [sip-101] ; Aastra 480i phones for general office type=peer insecure=very disallow=all allow=ulaw allow=alaw host=dynamic dtmfmode=auto canreinvite=no context=office-dial qualify=yes username=101 secret=xxxxxx mailbox=101 callerid="User 1" <101> sip show...

Hi, I'm getting this error when registering with SIP server using Asterisk 1.4.10 and Freepbx... I'm getting this error no matter what I try to setup in sip.conf : - I'm getting confused whether options are maxexpirey=36000 or maxexpiry=36000 ? - Can I solve this with some settings in sip.conf or is this problem harder ? - I've read something about Asterisk's bug on this error, but am not sure it really patching is necessary or can be avoided with different settings ? Thanks in advance, regards, R...

...my configuration information: sip.conf (I've tried with nat=no and it didn't help) [general] context=from-sip ; Default context for incoming calls port=5060 ; UDP Port to bind to (SIP standard port is 5060) bindaddr=0.0.0.0 ; IP address to bind to (0.0.0.0 binds to all) maxexpirey=3600 defaultexpirey=120 callerid=No CallID tos=lowdelay; 0x18 ; reliabile before dtmfmode=inband srvlookup=yes ;progressinband=no nat=yes notifymimetype=text/plain [broadvoice] type=friend username=801527xxxx (hid real number) fromuser=801527xxxx (hid real number) secret=xxxxxxxxxxxx (hid real pa...

Hi.. I just wondering why DTMF are not recognized by aterisk on incoming calls from my SIP provider ... ANy suggesteions ?` /Mike

...t to bind to bindaddr = 192.168.5.1 ; Address to bind to context = default ; Default for incoming calls ;srvlookup = yes ; Enable SRV lookups on outbound calls ;pedantic = yes ; Enable slow, pedantic checking for Pingtel ;tos=lowdelay ;tos=184 ;maxexpirey=3600 ; Max length of incoming registration we allow ;defaultexpirey=120 ; Default length of incoming/outoing registrati ;notifymimetype=text/plain ; Allow overriding of mime type in NOTIFY ;videosupport=yes ; Turn on support for SIP video ;disallow=al...